Hello everybody,
Do I understand correctly that Asterisk does not support sending INFO request?
Here is the goal I want to accomplish and I'd be happy to hear how can
it be done with asterisk. Asterisk needs to dial out and after
successful call establishment it needs to send in-dialog INFO
Hmmm... Folks, I beg you pardon, if I'm telling something which was said
before, but actually I have not found this anywhere, neither on
Voip-info.org or in several Asterisk's docs.
So, here is the statement:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them
will
Hello folks,
I've did some tests with different phones and Asterisk last two days and
here are some results, which I want to share with audience.
Cisco's 79xx and Sipura's phones/adapters on INVITE always reply with their
preferred codec.
So, for example, if Cisco's/Sipura's phone has
Hello!
Hmmm... Folks, I beg you pardon, if I'm telling something which was said
before, but actually I have not found this anywhere, neither on
Voip-info.org or in several Asterisk's docs.
So, here is the statement:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them
Hello Olle,
It is easier to turn it around:
Asterisk will issue a re-invite unless there is a reason
to keep the audio going through Asterisk
* NAT traversal issues
* Canreinvite=no
* Anything that needs asterisk to listen for DTMF in call
* Codecs that needs to be transcoded
Ok, let's dig into
Hello,
I know that using it is possible to dial several channels.
Question is - is it possible and if yes, how to dial several channels with
different ringing timeout?
I mean the following - for example when SIP/500 is dialed, I want three
phones to be dialed simultaneously - 1000, 2000 and
is works much
better.
regards
shams
On 6/14/05, Irakli Natsvlishvili [EMAIL PROTECTED] wrote:
Hello,
I have one question regarding *. Default configuration for asterisk is to
keep configuration(s) in ordinary text based config files.
My question is simple: is it possible to keep those config info
, but the transition and
integration with mysql was easy. Just store the asterisk specific data
in mysql, everthing else in ms sql if you must.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili
Sent: Tuesday, June 14, 2005
Hello,
I have one question regarding *. Default configuration for asterisk is to
keep configuration(s) in ordinary text based config files.
My question is simple: is it possible to keep those config info (at least,
to start from - sip.conf, extensions.conf and voicemail.conf) on a database,
When in realtime mode, does * uses static configs at all? Is it possible to
operate in realtime mode and have static configs (which are build based on
information taken from DB) as fallback solution?
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi there,
I need your help. Please le me know if it is possible to have following
implementation in place:
Asterisk server #1 (ast1) has server SIP clients with extensions 17XX
Asterisk server #2 (ast2) has server SIP clients with extensions 16XX
All I need that extensions from ast1 be able to
I've got * registered with 50 SIP extensions. There are two another SIP
proxies. I'd like to configure following:
1. Call from outside comes on *. * looks up for an extension
2. If no registered extension is on *, then request is forwarded to SIP
proxy 1.
3. If client in not found on SIP Proxy
Is there a way to display registered SIP useragents and sort them from CLI?
I.N.
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@home do that for you everyday...;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irakli
Natsvlishvili
Sent: Sunday, May 15, 2005 2:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question
Hello,
How do I
Hello,
Question #1:
I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905.
If g729 is the only available codec for 7905's configuration, then call from
7960 to 7905 goes without any problem and both phones use g729.
But if I call from 7905 to 7960 the following is displayed
Hello,
How do I routinely backup all necessary configuration files on [EMAIL
PROTECTED] Is
there any procedure/tool/script for it? And if I need to move * with
existing configuration on a new hardware, what is the best way to do it?
Thanks
I.N.
___
Hello everybody,
Further interesting details about BT-100, * and Cisco 7960.
Asterisk has G729 installed, on BT-100 there is g729 selected on all codec
selections. On Cisco 7960 preferred codec is g711.
Form sip.conf
[1707]
;- Cisco 7960
context=default
type= friend
username=1707
host
Only signaling or with media stream also?
You need commercial hardware platform. Those cost ~$20-100K. Probably you
can rent those boxes. I do know, that Spirent Communications has boxes for
SIP/H323/Skinny.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
What do you mean?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki
Sent: Tuesday, May 03, 2005 3:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Is there any chance to bring Skype
and
If there is another MoH source what is the correct way to use it with
extensions? Let me explain it - asterisk has MoH on extension 555.
Call comes on extension 111, so asterisk should connect incoming call to
extension 555 until someone answers on extension 111.
Second question: if there is a
This is an ordinary HP/Compaq/IBM server. You can install * on those servers
and install CCM on a ordinary computer with Intel chipset without much
problems.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Walid Azab
Sent: Thursday, April
Here is the situation. I've got * installed. I have Grandstream BT-100 with
latest beta firmware installed and Cisco 7960G.
[3710]
; - Grandstream
context=default
type=friend
username=3710
user=phone
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
qualify=2000
disallow=all
Hello everybody,
I'd like to know was there any load tasting done with *? Let's imagine 500
SIP clients on a server, 80 simultaneous calls. No transcoding, G711 or G729
codecs are used between endpoints.
How asterisk performs with 80 simultaneous calls when it sits on a media
stream? Is there
If there is another MoH source what is the correct way to use it with
extensions?
Thanks,
Irakli
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Hi there,
There are plenty of good documents on Asterisk, SIP and NAT on the
voip-info.org wiki. Please look them up. There are also information
within the configs/sip.conf.sample file within Asterisk.
Folks, let's face it - documentation on Asterisk sucks big time. This is
the reason why
On what trascoding time depends on?
I started server, run * and issued command show translations
--
sipsrv1*CLI show translation
Translation times between formats (in milliseconds)
Source Format (Rows)
Two questions.
If there is a VoIP-VoIP call, how do I see from a console what codecs are in
use by peers?
Second question: if there is transcoding going on, how do I see detailed
information about it - peers involved, extensions, IP addresses, ports,
codecs from/to and so on?
Thanks,
Irakli
100k question - does asterisk correctly handle following situations:
1. Asterisk is on a public IP
Two SIP clients on separate networks, each of them are behind dynamic NAT
gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought
asterisk.
2. Even worst case - three
Hello there,
How do I configure any type of action based caller's extension and dialed
number? For example if someone on extension 1777 calls extension 1777 this
should be treated as accessing his voicemail box, so he won't need to call
voicemail and entering mailbox number and password.
I.N.
Hello, Alejandro!
AG I have a problem with ATA-186 configured for silence supression
Don't!
I.N.
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Hello, Matt!
MR fine. If you have to do any sort of transcoding a soekris is not the
MR way to go but for a small installation it works great.
Well.. Cisco's 17xx series router is a device which you can take, plug,
configure and have office PBX. But price tag is $2K.
Why the same can't be done
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, April 03, 2005 3:33 AM
Subject: [Asterisk-Users] Router with QoS recommendations
As I have a Cisco PIX 515, with NO QoS functionality,
and I'm looking for a router that does outgoing QoS to put in front of my
PIX.
PixOS
I don't know following has debated here or not, but is there in this world
following stuff:
A small, physically small box, like cable/DSL router, which comes with:
1) Ethernet port, 2) Console port, 3) CompactFlash or USB port, 4) memory
module port, like SODIMM
Box has built-in flash (256MB
Hello, Olle!
OEJAsterisk 2.0 was moved to a Microsoft platform due to the
OEJdemand for higher stability and a more secure foundation.
Nice...
I remember that about 10 years ago, when I was working in a daily newspaper
we wrote and article on April 1st on a first page about scientific
Any idea when this gonna be fixed?
-Original Message-
From: Kanuri, Seshu (Company IT)
[mailto:[EMAIL PROTECTED]
Sent: Thursday, March 31, 2005 7:20 AM
To: [EMAIL PROTECTED]; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem with
Rule of thumb - echo is caused by remote node. Check other end.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Philip Siegrist
Sent: Thursday, March 31, 2005 7:37 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Echo on
If you are using FC4, I think you are using Kernel 2.6, in which case
usb is not needed.
Anyway, is there a cure for this problem?
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Hello everybody,
I've run on a problem with music on hold. Asterisk does not play anything.
Here is the info:
latest Asterisk:
Asterisk CVS-HEAD-03/25/05-23:18:57
Asterisk is installed Fedora Core 4 running on AMD 2.0Ghz CPU box with 512
RAM.
I took latest zaptel source code, uncommented
G'day mate,
I've got 15 7960/7940 in my office with firmware 7.4 and have no problems.
I can make calls from the 7960. When I get a call placed to the 7960 the
call
is setup but there is no audio in either direction.
Is call from 7960 to 7960?
I have tried firmware versions 6 7 on the Cisco
ntpdate ntp1.cs.wisc.edu
30 Mar 23:15:20 ntpdate[3840]: no server suitable for synchronization found
time-a.nist.gov
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